From e273695cc90f6dea5399c499603e4425be261ba0 Mon Sep 17 00:00:00 2001
From: John Preston <johnprestonmail@gmail.com>
Date: Fri, 8 Dec 2017 12:57:43 +0400
Subject: [PATCH] Fix planar audio playback (for example .flac).

Also add some more crash information logging.

Fixes #4120.
---
 Telegram/SourceFiles/logs.cpp                 |  3 ++
 .../media/media_audio_ffmpeg_loader.cpp       | 48 ++++++++++++++++---
 2 files changed, 45 insertions(+), 6 deletions(-)

diff --git a/Telegram/SourceFiles/logs.cpp b/Telegram/SourceFiles/logs.cpp
index 5db94cafc..b344ae046 100644
--- a/Telegram/SourceFiles/logs.cpp
+++ b/Telegram/SourceFiles/logs.cpp
@@ -1139,6 +1139,9 @@ namespace internal {
 	}
 
 	void setCrashAnnotation(const std::string &key, const QString &value) {
+		static QMutex mutex;
+		QMutexLocker lock(&mutex);
+
 		if (!value.trimmed().isEmpty()) {
 			internal::ProcessAnnotations[key] = value.toUtf8().constData();
 		} else {
diff --git a/Telegram/SourceFiles/media/media_audio_ffmpeg_loader.cpp b/Telegram/SourceFiles/media/media_audio_ffmpeg_loader.cpp
index f29b89beb..fec669f74 100644
--- a/Telegram/SourceFiles/media/media_audio_ffmpeg_loader.cpp
+++ b/Telegram/SourceFiles/media/media_audio_ffmpeg_loader.cpp
@@ -20,10 +20,23 @@ Copyright (c) 2014-2017 John Preston, https://desktop.telegram.org
 */
 #include "media/media_audio_ffmpeg_loader.h"
 
+namespace {
+
 constexpr AVSampleFormat AudioToFormat = AV_SAMPLE_FMT_S16;
 constexpr int64_t AudioToChannelLayout = AV_CH_LAYOUT_STEREO;
 constexpr int32 AudioToChannels = 2;
 
+bool IsPlanarFormat(int format) {
+	return (format == AV_SAMPLE_FMT_U8P)
+		|| (format == AV_SAMPLE_FMT_S16P)
+		|| (format == AV_SAMPLE_FMT_S32P)
+		|| (format == AV_SAMPLE_FMT_FLTP)
+		|| (format == AV_SAMPLE_FMT_DBLP)
+		|| (format == AV_SAMPLE_FMT_S64P);
+}
+
+} // namespace
+
 bool AbstractFFMpegLoader::open(qint64 &position) {
 	if (!AudioPlayerLoader::openFile()) {
 		return false;
@@ -384,16 +397,33 @@ AudioPlayerLoader::ReadResult FFMpegLoader::readFromReadyFrame(QByteArray &resul
 		// frame->channels = 1
 		//
 		// So the frame just wasn't consistent with the codec params.
-		if (fmtContext->streams[streamId]->codecpar->channels > 1
-			&& frame->extended_data[1] == nullptr) {
-			auto params = fmtContext->streams[streamId]->codecpar;
-			LOG(("Audio Error: Inconsistent frame layout/channels in file, codec: (%1;%2), frame: (%3, %4)"
+		if (frame->extended_data[1] == nullptr) {
+			const auto params = fmtContext->streams[streamId]->codecpar;
+			if (IsPlanarFormat(params->format) && params->channels > 1) {
+				LOG(("Audio Error: Inconsistent frame layout/channels in file, codec: (%1;%2;%3), frame: (%4;%5;%6)."
+					).arg(params->channel_layout
+					).arg(params->channels
+					).arg(params->format
+					).arg(frame->channel_layout
+					).arg(frame->channels
+					).arg(frame->format
+					));
+				return ReadResult::Error;
+			} else {
+				const auto key = "ffmpeg_" + std::to_string(ptrdiff_t(this));
+				const auto value = QString("codec: (%1;%2;%3), frame: (%4;%5;%6), ptrs: (%7;%8;%9)"
 				).arg(params->channel_layout
 				).arg(params->channels
+				).arg(params->format
 				).arg(frame->channel_layout
 				).arg(frame->channels
-				));
-			return ReadResult::Error;
+				).arg(frame->format
+				).arg(ptrdiff_t(frame->data[0])
+				).arg(ptrdiff_t(frame->extended_data[0])
+				).arg(ptrdiff_t(frame->data[1])
+				);
+				SignalHandlers::setCrashAnnotation(key, value);
+			}
 		}
 
 		if ((res = swr_convert(swrContext, dstSamplesData, dstSamples, (const uint8_t**)frame->extended_data, frame->nb_samples)) < 0) {
@@ -401,6 +431,12 @@ AudioPlayerLoader::ReadResult FFMpegLoader::readFromReadyFrame(QByteArray &resul
 			LOG(("Audio Error: Unable to swr_convert for file '%1', data size '%2', error %3, %4").arg(_file.name()).arg(_data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
 			return ReadResult::Error;
 		}
+
+		if (frame->extended_data[1] == nullptr) {
+			const auto key = "ffmpeg_" + std::to_string(ptrdiff_t(this));
+			SignalHandlers::setCrashAnnotation(key, QString());
+		}
+
 		int32 resultLen = av_samples_get_buffer_size(0, AudioToChannels, res, AudioToFormat, 1);
 		result.append((const char*)dstSamplesData[0], resultLen);
 		samplesAdded += resultLen / sampleSize;